Displaying 9 results from an estimated 9 matches for "nortpproxy".
2014 Jun 18
1
Making sense of SDP for debugging of missing audio in SIP trunk
...(inside 200 OK response body from asterisk)
v=0
o=root 835643920 835643920 IN IP4 201.234.196.171
s=Asterisk PBX 11.10.0
c=IN IP4 201.234.196.171
t=0 0
m=audio 12112 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
a=nortpproxy:yes
In contrast, all failing calls have SDP negotiation that looks like this:
(inside INVITE request body from SIP carrier)
v=0
o=msw.chance4minutes.net 1234 0 IN IP4 38.126.208.46
s=sip call
c=IN IP4 38.126.208.46
t=0 0
m=audio 30534 RTP/AVP 0 18
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmt...
2007 Jul 17
2
media not accpetable with outgoing call on cisco
...ing;id-type=subscriber;screen=yes
Content-Type: application/sdp
v=0
o=MxSIP 0 198 IN IP4 192.168.0.249
s=SIP Call
c=IN IP4 200.200.100.106
t=0 0
m=audio 39318 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=direction:active
a=nortpproxy:yes
*Jul 17 15:57:02.608: 0x64C01D00 : State change from (STATE_NONE,
SUBSTATE_NONE) to (STATE_IDLE, SUBSTATE_NONE)
*Jul 17 15:57:02.608: sipSPIStreamTypeAndDtmfRelay: ERROR - no voice codec
and no dtmf-relay match
*Jul 17 15:57:02.608: sipSPIDoAudioNegotiation: Media negotiation failed for
m-lin...
2005 Oct 10
1
Incoming SIP getting in, but not ringing.
...2 4 5 110 7 10
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:110 speex/8000
a=rtpmap:7 LPC/8000
a=rtpmap:10 L16/8000
a=silenceSupp:off - - - -
a=direction:active
a=nortpproxy:yes
17 headers, 20 lines
Using latest request as basis request
Sending to 217.10.79.219 : 5060 (non-NAT)
Found peer 'SipGate'
Reliably Transmitting (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 217.10.79.219;branch=z9hG4bKeafc.0df1fb45.0
Via: SIP/2.0/UDP 217.10.79.8;...
2005 Oct 09
1
Problem setting SIP incoming/outgoing
...<http://109.147.41.48>
t=0 0
m=audio 53870 RTP/AVP 0 8 18 3 101
c=IN IP4 109.147.41.48 <http://109.147.41.48>
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=direction:passive
a=nortpproxy:yes
--- (26 headers 16 lines)---
Using INVITE request as basis request -
FADDE365-357711DA-80F5C727-E0F535F0@209.47.41.61
Sending to 109.147.41.48 <http://109.147.41.48> : 80 (non-NAT)
Found peer 'sipserverout'
Reliably Transmitting (no NAT) to 209.47.41.48:80 <http://209.47.41.48...
2009 Sep 28
0
Asterisk complaning about no such host -- never asked to contact the host it complains about
...p:4444444 at 10.252.1.7>;reason=unconditional
v=0
o=root 1273543599 1273543599 IN IP4 10.252.1.7
s=Asterisk PBX 1.6.0.10
c=IN IP4 10.252.1.7
t=0 0
m=audio 39110 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=nortpproxy:yes
<------------->
--- (17 headers 13 lines) ---
[Sep 28 15:41:24] WARNING[32316]: chan_sip.c:4224 create_addr: No such
host: 5555555
Why would it be trying to contact the host 5555555? There's nothing in
the invite indicating that as a host. Furthermore, the verbosity level
was at the...
2009 Dec 07
1
Error : SIP/2.0 401 Unauthorized
Hi Friends,
need to help.
*I have problem about sip : SIP/2.0 401 Unauthorized*
Is it require to nathelper module in kamailio ?
*what can i write kamailio.cfg file when kamailio and Asterisk on same
network?*
Scenario is like as :
-----------------------------
1) kamailio server on 172.18.100.74
kamailio.cfg ( nathelpler module )
-----------------
loadmodule "nathelper.so"
2012 Jun 18
1
Error SIP/2.0 488 Not acceptable here
...MU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:99 G726-32/8000
> a=rtpmap:108 AAL2-G726-32/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:20
> a=sendrecv
> a=direction:active
> a=nortpproxy:yes
> <------------->
> [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: --- (25 headers 21 lines) ---
> [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Sending to 217.10.79.9:5060(NAT)
> [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Using INVITE request as basis
> request - 4fdf703d880d-...
2013 Apr 09
1
[OpenSIPS-Users] 404 When BYE initiated by external callee
...UNKNOWN__and_probably_unsupported.
> c=IN IP4 192.168.2.5.
> t=0 0.
> m=audio 31148 RTP/AVP 18 101.
> a=rtpmap:18 G729/8000.
> a=fmtp:18 annexb=no.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=silenceSupp:off - - - -.
> a=ptime:20.
> a=sendrecv.
> a=nortpproxy:yes.
>
>
>
> U 2013/04/09 12:17:06.988918 108.59.2.133:5060 -> 192.168.2.5:5060
> BYE sip:1001 at 70.10.163.44:5060 SIP/2.0.
> Max-Forwards: 64.
> To: "1001" <sip:1001 at 70.10.163.44>;tag=as4b40d9b4.
> From: <sip:001110215178342008 at sbc.voxbeam.com&...
2005 Sep 29
1
Cisco AS5300 --> [SIP] --> Asterisk - NO AUDIO
...0.237
t=0 0
m=audio 37708 RTP/AVP 18 4 3 8 0 110
c=IN IP4 203.88.192.42
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:110 X-NSE/8000
a=fmtp:110 192-194
a=direction:passive
a=direction:active
a=nortpproxy:yes
--- (24 headers 19 lines)---
Using INVITE request as basis request -
805AF00B-305C11DA-814CFCF5-33432EF@211.147.240.237
Sending to 203.88.192.42 : 5160 (non-NAT)
Found no matching peer or user for '203.88.192.42:5160'
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio f...